Re: Streaming setup with OBS and JACK

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



On Wed, 25 Jul 2018, David Kastrup wrote:

Paul Davis <paul@xxxxxxxxxxxxxxxxxxxxx> writes:

On Wed, Jul 25, 2018 at 2:56 AM, <hollundertee@xxxxxxx> wrote:

2) Is there a reliable and if possible hassle free way to get ALSA and
PA into JACK? ALSA loopback would probably not work for programs using
PA directly.

​it works just fine. I use it with Skype with no issues.​

For pulseaudio/jack, I use the following "pulsejack" script.  Without
arguments, it tries connecting a running jackd with a running pulseaudio
system.  With arguments, it kills any existing jackd, then starts a new
one with the given arguments and connects it with a running pulseaudio
system.  I'd like it to use things like jack_wait and similar in order
to avoid the clumsy manual job management but haven't been able to make
this robust enough (namely without having it lock up or not work in some
situations).

I have been using PA and Jack bridged for about 4 years now. My script has changed during that time, not for problems but to make it more generic. I use jackd as the audio device for pulse. That is I remove the alsa and udev modules from pulse (after starting). This ensures that jack is always the default i/o and as a side benefit allows jack to freewheel with no problems and gives better xrun immunity. I do not use the jackdbus-detect module, but rather use jacksink or jacksource modules as needed to ensure separate stereo pairs rather 4 channels being used as suround... I always set the pulse modules _not_ to auto connect, prefering to have jack_connect, jack_disconnect or jack-plumbing do that job to the port I want connected (system_1/2 are not always right).

I use this for all desktop audio, all the time and have found this to be a robust solution. I have also set up a systray button that drops the PA-jack bridge for times I wish to use a lower latency than pa does well (or some of the pa clients like skype that don't handle low latency at all)

There is very little manual anything after configuration. Both jack and pulse are started automatically at session start and run till session end. The point of jack in the first place is to allow routing changes on the fly in a manual way, auto connect just gets in the way of that except for reconnecting things when restarting a jack client to where they were before it stopped.

I use zita-ajbridge if I must have more than one audio device. The truth about this is that I use more than one audio device just to make sure that it works and so that I can help other people get their USB mic working. I have only one audio interface worth using for anything where quality matters: my delta 66 a semi-pro audio interface that is long obsolete. I have also found that I don't use more than 2 channels at a time anyway so even 6 i/o has been overkill for me.

--
Len Ovens
www.ovenwerks.net
_______________________________________________
Linux-audio-user mailing list
Linux-audio-user@xxxxxxxxxxxxxxxxxxxx
https://lists.linuxaudio.org/listinfo/linux-audio-user

[Index of Archives]     [Linux Sound]     [ALSA Users]     [Pulse Audio]     [ALSA Devel]     [Sox Users]     [Linux Media]     [Kernel]     [Photo Sharing]     [Gimp]     [Yosemite News]     [Linux Media]

  Powered by Linux